Telecommunication operators are migrating their wireline and wireless networks from the existing Circuit Switched (CS) technology to an IP Multi media System (IMS). IMS has a richer call control capability than CS. With IMS, the call establishment may, for example, be enriched with multi media alerting information or caller information. IMS may be used for establishing speech calls (voice session), but also for non-call related activities, such as establishing a chat session or for transferring a file or the like.
Deployment of IMS in wireline networks is leading the deployment of IMS in wireless networks. Reason is that the radio interface in wireless networks has inherently less transmission capability than wireline connections. Voice sessions require data connection that offers “Conversational Voice” grade Quality of Service (QoS), in order to ensure low transmission latency and sufficient speech quality. For non-call related activities, the QoS grade is less critical.
The Universal Mobile Telephony System (UMTS) may be suitable for IMS based voice sessions. However, in practice, High Speed Downlink Packet Access (HSDPA) and High Speed Uplink Packet Access (HSUPA), jointly referred to as High Speed Packet Access (HSPA), may be needed to offer conversational voice QoS for a large number of subscribers.
The current mobile communications networks like the Global System for Mobile communications (GSM) and UMTS Radio Access Networks (RAN) networks do, however, have sufficient capacity for offering CS based speech connection to a large number of users.
Technologies as Wireless Local Area Network (WLAN) and the Worldwide interoperability for Microwave Access (WiMAX) offer high data transmission speeds for large numbers of subscribers. Hence, IMS speech calls, including call enrichments, may be offered through WLAN and WiMAX. However, deployment of WLAN and WiMAX will remain limited to designated spots in urban areas. Therefore, mobile users will be compelled to revert to UMTS when outside urban areas.
As a result, large scale deployment of IMS for mobile speech calls is expected not to materialise before 2008-2010. And as an aggregate result, rich call experience will not be available until then.
To overcome the above-sketched dilemma, telecommunication operators are seeking mechanisms to combine the best of both worlds: that is use IMS for (call-related) data services and use CS for speech calls. Mechanisms to combine IMS and CS, to offer enriched call experience, are known as Combinational Services. Details of Combinational Services are specified in, amongst others, the Technical Specification of the 3rd Generation Partnership Project (3GPP), 3GPP TS 23.279. Combinational Services are also known as “combination of CS and IMS services” (CSI).
A dilemma with CSI is that it remains a CS call, enriched with IMS. The service network needs to remain geared for handling the CS call. A service like IMS Multi Media Office (IMS MMO), which may be considered VPN for IMS network, would need to be aware that a served subscriber is a GSM subscriber, for example, and would have to act accordingly.
Various initiatives have started to make the access network transparent for the service network. That concept enables a network service to be “access independent”. It could treat all calls as “IMS calls”. If the access used for a call is non-IMS, e.g. GSM, then a designated border access gateway would need to apply suitable conversion between IMS signalling and, in this example, GSM signalling. However, due to significant differences between GSM signalling and IMS signalling, it is practically not possible to provide full access transparency towards an IMS service.
3GPP has drafted various network CSI scenarios, known as Alpha, Beta and Gamma.
The Alpha solution entails that a CS call is combined with a PS-based media session. The CS call is used for the speech connection between the calling and called party. The PS session is used for multimedia transfer. The user terminals of the calling and called party correlate the speech connection and the media session and combine these into an enriched call. The Alpha solution is standardised in 3GPP as Combinational Services.
Quintessential to the Alpha solution is that the voice call between the calling and called party is CS end-to-end. The calling party uses the Mobile Subscriber (MS) Integrated Services Digital Network (ISDN) Number (MSISDN) of the called party to set up the call. The IMS signalling between the calling and the called party runs independently of the CS call signalling between the calling and the called party.
The Beta solution entails that a user has a Session Initiation Protocol (SIP) capable terminal. When the user establishes a call, the SIP User Agent uses SIP signalling for call establishment. A control node in the IMS network establishes a CS call towards the calling party; the CS call is also connected to a MediaGateWay (MGW), for connection to the called party for this call.
Inherent complexity of the Beta solution is that the establishment of the voice call is done under control of the IMS server (i.e. SIP Application Servers) (i.e. the “Beta server”). This makes the architecture more complex. The SIP-AS has to control two SIP Sessions for a multimedia voice call establishment:                a first SIP session is established by the calling party and traverses the Beta server; the Beta server needs to be aware that this calling subscriber (or the called subscriber) requires special handling for the establishment of the speech path;        another SIP session is established for the speech path. The Beta server initiates a SIP session towards the calling party, addressing her same by MSISDN. This SIP session will be converted to ISDN User Part (ISUP) signalling and Direct Transfer Application Part (DTAP) signalling, so it can be offered to the calling subscriber. This SIP session is used to get a voice call established from a MGW to the calling party.        
The Beta server then also establishes an additional SIP session towards the called party. This SIP session will also be converted to ISUP and DTAP, so it can be offered to the called party. The voice connection towards the called party will run from the MGW. Hence, voice between calling and called party is established through a centralised MGW.
The Beta server has to take additional steps to establish the speech path between the calling and the called party. This implies, amongst others, that the Beta server needs to have the MSISDNs of the calling and called party. The establishment of the CS call to the mobile calling and called party entails also ISUP routing through a Gateway Mobile Switching Centre (GMSC) and the Mobile Switching Centre (MSC), Home Location Register (HLR) interrogation etc. A possible terminating call Intelligent Network (IN) service for the calling or called party needs to be suppressed for this call. This makes Beta a complex solution.
The Gamma solution entails that a GSM terminal is connected to the GSM network as per current methodology. When the user initiates a call, standard CS signalling is used, i.e. the terminal uses DTAP signaling towards the serving MSC. The MSC will convert the DTAP signalling to the SIP signalling. The MSC has built-in Media Gateway Control Function (MGCF), enabling the MSC to route the call immediately to the IMS network. The MSC emulates an IMS User Agent, on behalf of the GSM user. When the user establishes a call, this call will be offered to the IMS network as originating IMS call.
Quintessential to the gamma solution is that the call establishment from the terminal is CS. The calling party uses the MSISDN of the called party to establish the call. A voice connection, for example, between calling and called party is established through CS signaling form the terminals.
Although SIP signaling may be present in the Gamma solution this is, however, not related to the establishment of the voice call. Such SIP signaling may be used for the transfer of multimedia components between the calling and called party.
Comparing the Alfa, Beta and Gamma solutions shows that neither thereof provides a fully transparent IMS call establishment end-to-end.